Html web rtc
- 1. Web RTC
- 2. Web RTC introduced by World Wide Web Consortium (W3C). That supports browser-to browser applications for voice calling, video chat, and P2P file sharing. If you want to try out? web RTC available for Chrome, Opera, and Firefox. A good place to start is the simple video chat application at here . Web RTC implements three API's as shown below . MediaStream - get access to the user's camera and microphone. RTCPeerConnection - get access to audio or video calling facility. RTCDataChannel - get access to peer-to-peer communication.
- 3. MediaStream The MediaStream represents synchronized streams of media, For an example, Click on HTML5 Video player in HTML5 demo section or else click here . The above example contains stream.getAudioTracks() and stream. VideoTracks(). If there is no audio tracks, it returns an empty array and it will check video stream,if webcam connected, stream.getVideoTracks() returns an array of one MediaStreamTrack representing the stream from the webcam. A simple example is chat applications, a chat application gets stream from web camera, rear camera, microphone. Sample code of MediaStream function gotStream(stream) { window. AudioContext = window. AudioContext window. webkitAudioContext; var audioContext new AudioContext(); mediaStreamSource . connect (audioContext . destination) navigator . getUserMedia ( {audio : true} gotStream) ;
- 4. Screen capture It's also possible in Chrome browser with mediaStreamSource and it requires HTTPS. This feature is not yet available in opera. Sample demo is available at here Session Control, Network & Media Information Web RTC required peer-to-peer communication between browsers. This mechanism required signalling, network information, session control and media information. Web developers can choose different mechanism to communicate between the browsers such as SIP or XMPP or any two way communications. A sample example of XHR is here. Sample code of createSignaIingChannel()
- 5. var signalingChanne1 createSigna1ingChanne1() var pc; var configuration // run start(true) to initiate a call function start(isCa11er) { pc new RTCPeerConnection(configuration); // send any ice candidates to the other peer pc . onicecandidate function evt) { signalingChanne1.send(JSON. "candidate" : evt . candidate // once remote stream arrives, show it in the remote video element pc . onaddstream function (evt) { remoteView. s rc URL . createObjectURL (evt . stream) ; // get the Local stream, show it in the Local video element and send it navigator.getUserMedia({ "audio" : true, "video" : true } , function (stream) { selfView. src URL. createObjectURL(stream); pc . addStream(stream) if (isCa11er) pc . createOffer (gotDescription) else pc . createAnswer(pc . remoteDescription, gotDescription) function gotDescription(desc) { pc . setLoca1Description (desc) signalingChanne1. send (JSON . stringify({ " sdp" : desc signalingChanne1. onmessage function (evt) { if ( ! pc) start (false) var signal JSON . parse(evt . data) if (signal.sdp) pc. setRemoteDescription(new RTCSessionDescription(signa1.sdp else pc . addlceCandidate(new RTCIceCandidate(signa1. candidate));